Method of digitally evaluating the frequency and the phase of signals, and a device for implementing such a method

ABSTRACT

A method of digitally evaluating the frequency and the phase of signals in the form of digitized samples, the method being wherein it comprises in succession: 
     a stage during which numbers corresponding to the samples of the signal to be analyzed are processed in order to convert them into the form of an analytic signal whose real portion coincides with said signal to be analyzed; and, in parallel therewith: 
     a stage of estimating the parameters to be analyzed which is performed in an overall manner on the basis of estimators and of selection criteria by working on the phase of the signal without using any operator of the Fourier transform type or any hypothesis test either separately or simultaneously; and 
     a stage of estimating the differences between the real signal as taken in this way and the signal obtained from the estimated parameters, thereby making it possible to deliver data in digital form relating to the quality of the analyzed signal and to the reliability of the estimated values. The invention is applicable, in particular, to microwaves.

The invention relates to a method of digitally evaluating the frequencyand the phase of signals, and to a device for implementing such amethod.

BACKGROUND OF THE INVENTION

As in a conventional frequencymeter or phasemeter, it serves toreconstitute respectively the frequency or the phase of the signal to beevaluated, and to provide an indication of the accuracy with which theseparameters are measured together with items for determining the qualityof the signal (e.g. signal/noise ratio, form factor).

The signal analysis methods commonly used for measuring frequency andphase are generally based either:

on analog processing and filtering methods for performing interation orcounting zero crossing, with the duration of the integration or thefiltering being a function of the desired accuracy or:

on methods for locking an oscillator, for example by means of a phaselock loop, thereby isolating the signal to be measured so thatmeasurements can then be performed thereon; or else;

directly on the signal from the locked oscillator, or on the controlsignals thereof.

The use of these methods suffers from the drawback of not storing theresults of earlier processing. As a result, any parameter valueestimated from one portion of the signal, is actually made use of inprocessing a different portion of the signal subsequent to the portionwhich was used for obtaining the initial estimate, and this hasunfortunate consequences both:

on performance, since this technique cannot provide the best possibleadaptation to variability in the parameters, and as a result it is moresensitive to disturbing signals (noise, interfering spectrum lines)superposed on the signals to be measured; and also

on complexity, since when using the above-described techniques, it isvery difficult to provide optimum adaptation at each of the stages thatneed to be implemented in order to achieve the desired aim. Inparticular, the stages of acquisition and of tracking require differentcharacteristics, so they either separate devices, or else they requirethe characteristics of a set of analog circuits to be adapted to each ofthese stages, and thus without excessive complication, it is verydifficult for the measuring device to be optimally adapted and give bestresults for each of these stages.

Further when performing successive estimates of the same parameter, itis not possible to use the same signal. The results of successiveestimates are thus performed over signal periods taken successively intime, which makes it essential to use hypotheses which are guaranteed toprovide valid and reproducible results.

The object of the present invention is to mitigate these drawbacks.

SUMMARY OF THE INVENTION

For this end, the present invention provides a method of digitallyevaluating the frequency and the phase of signals in the form ofdigitized samples, wherein the method comprises in succession:

a stage during which numbers corresponding to the samples of the signalto be analyzed are processed in order to convert them into the form ofan analytic signal whose real portion coincides with said signal to beanalyzed; and, in parallel therewith:

a stage of estimating the parameters to be analyzed which is performedin an overall manner on the basis of estimators and of selectioncriteria by working on the phase of the signal without using anyoperator of the fourier transform type or any hypothesis test eitherseparately or simultaneously; and

a stage of estimating the differences between the real signal as takenin this way and the signal obtained from the estimated parameters,thereby making it possible to deliver data in digital form relating tothe quality of the analyzed signal and to the reliability of theestimated values.

These various stages are advantageously provided by means of a set ofequipment comprising a digitizer and a calculating member, suitable foraccepting a wide variety of signals and for delivering the desiredparameters.

Advantageously, in the stage for estimating the parameters to beanalyzed, the phase of the analytic signal is replaced by its associateddeveloped phase.

It is possible to use unbiased estimators with minimum variance in themethod of the invention, thereby making it possible to optimize theduration of the processed signal.

Further, the estimate of the degree of conformity in the analyzed signalmakes it possible to eliminate jamming, in particular when the signal tobe analyzed is fleeting. This makes it possible to consider extractingthe parameters characteristic of a short signal, assuming that theprobable instant of its arrival is known, or to determine the instant atwhich a signal of known characteristics, does in fact, appear. Forexample, in the case of a message which is transmitted periodically orrandomly, the method may be used either:

to determine the precise instants at which the message begins, therebymaking it possible subsequently to extract its characteristics; or else

if the message has characteristics which are accurately known, andarrives at random instants in time, to determine thebeginning-of-message instants.

More precisely, in the method of the invention, the initial samples arestored in a memory from which they are extracted as often as may berequired by each new calculation giving rise, to within given accuracy,to the signal/noise ratio of the analyzed signal, to the duration of thesignal to be taken into account, and to the accuracy of the phasemeasurement and of the frequency measurement, with the above being basedon a given quality parameter.

Such a method may be applied to sinusoidal signals having noise orharmonics due to non-linearities added thereto.

It may be applied to signals of arbitrary shape (e.g. square,triangular, or Gaussian), for the purpose of determining accurate shapecharacteristics in addition to the frequency and the phase.

The invention also relates to a device for implementing the method.

This device, for example a frequencymeter, comprises a digital processormember having an input to which the signal to be analyzed is appliedafter passing through a digitizer, said digitizer also receiving astable digital clock signal, with the output from the digital processormember providing a quality signal and a frequency measurement signal.

Advantageously, this digital processing member comprises:

a common memory circuit;

a circuit for generating the analytic signal;

a circuit for calculating samples of the phase of the analytic signaland for constructing the developed phase;

a circuit for determining the quality of the signal and enabling anestimate to be made of the statistics of the differences from theestimated sinewave and the estimated noise as a function of the form ofthe signal to be analyzed;

a circuit for estimating frequency;

with all of the above circuits being connected via respective both-waylinks to an interconnection bus;

an input circuit for acquiring digitized samples, said circuit beingconnected to the common memory via a one-way link; and

a coupler having its inputs connected to the outputs from the circuitfor determining the quality of the signal and from the circuit forestimating its frequency, and whose output constitutes the output fromsaid processor member.

This device could also be a phasemeter, in which case it would include asecond digitizer receiving firstly a phase reference signal and secondlythe stable reference clock signal, and the processing member wouldfurther comprise:

a second circuit for acquiring digital samples of the phase referencesignal and connected like the first sample acquisition circuit to thecommon memory circuit; and

a circuit for estimating the phase relative to the reference signal, andfor estimating the jitter.

Advantageously, these devices are physically constituted by using ahardware structure comprising a plurality of mutually independentcalculation units which share the various calculation tasks required forthe analysis and which have access to the memory means which serve tointerconnect said calculation means, said addressable memories beingeither common (bus or multiport), or else constituted as a latticenetwork, with an exclusive zone being attributed to each calculationmember for writing in memory.

BRIEF DESCRIPTION OF THE DRAWINGS

Embodiments of the invention are described by way of example withreference to the accompany drawings, in which:

FIGS. 1 to 4 are diagrams showing the operation of constructing thedeveloped phase; and

FIGS. 5 and 6 are block diagrams of two devices for implementing themethod of the invention.

MORE DETAILED DESCRIPTION

The method of the invention comprises:

A stage in which the signal to be analyzed is transformed into digitalsamples. These samples are taken directly from the signals to beprocessed and are stored in an addressable digital memory. Methods ofdigitizing a signal, and of storing the digital samples are known. Theycan be implemented using architectures based on components which arecommercially available.

A stage in which the numbers obtained in this way and representative ofthe signal to be analyzed are processed in order to put them into theform of an analytic signal whose real part coincides with the originalsignal.

A stage during which the parameters to be analyzed are estimated on thebasis of estimators and selection criteria operating on the phase of theanalytic signal (modulo ±π) which is replaced by its associateddeveloped phase in order to simplify processing.

The term "modulo ±π" is used throughout this specification, by analogywith "modulo n" to have the following meaning:

    (Φ.sub.i =calculated phase; Φ=phase modulo ±π)

    if -π≦Φ.sub.i <+π then Φ=Φ.sub.i

    if +π≦Φ.sub.i then Φ=Φ.sub.i -2kπ

    if Φ.sub.i <-π then Φ=Φ.sub.i +2kπ

with k being a positive integer chosen so that -π≦Φ<+π.

A stage for estimating the difference between the real signal as definedabove and the signal obtained from the estimated parameters, therebymaking it possible to deliver data in digital form relating to thequality of the analyzed signal and also to the reliability of theestimated values (form factor, additive noise, interfering spectrumlines), with these judgment criteria naturally be a function of thecharacteristics of the estimator used.

Background required for understanding the method and relating to theanalytic signal and the developed phase are recalled below.

The method of the invention is based on using mathematical principleswhich are known elsewhere and whose results are recalled below, and onan original method of constructing the developed phase of a signal whichis described below.

The concept of an analytic signal associated with a real signal r(t) wasintroduced by Ville. This concept, limited to the case of "finiteenergy" signals (which corresponds to the kinds of signal that arenormally processed), serves to associate a real signal r(t) with asecond function q(t) of the same type such that the complex function:

    x(t)=r(t)+jq(t)

has a spectrum without any negative frequency components.

The function x(t) is the "analytic signal" and the function q(t) is the"quadrature signals", both of which are associated with the givenfunction r(t).

Thus, for a sinusoidal signal:

    r(t)=A cos(2πvt+φ)=real part of A exp[j(2πvt+φ)]

    q(t)=A sin(2πvt+φ)=imaginary part of A exp[j(2πvt+φ)]

where x(t)=A exp[j(2πvt=φ)]

This concept, when applied to finite energy real signals possessing afourier transform, provides a method of calculation for deducing thequadrature signal form the real signal r(t), and thus of associatingwith r(t) the complex function x(t) having no negative frequencycomponents, thereby making it possible to isolate the amplitude and thephase of the corresponding signal.

Thus, if: ##EQU1## where R(v) is the fourier transform of r(t), thusmaking it possible to calculate: ##EQU2## where Q(v) is the fouriertransform of q(t) such that:

    Q(v)=-jR(v) if v ≠0

    and x(t)=r(t)+jq(t)

In order to generate the analytic signal in demodulation and frequencytransposition equipment, the signal r(t) is mixed in conventional mannerboth with a sinewave from a local oscillator at an appropriatefrequency, and with a sinewave which is offset by π/2 relative to thefirst.

The result of this operation is to provide two signals which are derivedfrom r(t), namely a first signal r'(t) which is a mere transposition ofr(t) in the frequency spectrum, and a second signal q'(t) which is inquadrature with of r'(t). Such that X'(t)=r'(t)+jg'(t) fully representsall of the characteristics of r(t).

It is then possible to use a digital method to produce a signal such asa sinewave for generating the above-defined analytic signal.

FIG. 1 is a diagram representing a sinewave signal containing noise inthe real signal, where:

    r(t)=A cos(wt+Φn(t)

w=angular frequency

Φ=phase

n(t)=relative noise.

This signal may be associated with the analytic signal written asfollows:

    X(t)=A exp[j(wt+Φ)]+n*(t) with <|n*(t)|.sup.2 >=2σ.sup.2

where <>designates the mean value of the quantity between thearrowheads.

FIG. 4 is a cylindrical representation of an analytic signal. It showsthe noise-free signal 11, the cylindrical envelope 12 of the signal, theenvelope 13 of noise σ, and the value 14 of n(t). The power of theuseful signal is A² /2 and the noise variance is:

    σ.sup.2 =<n.sup.2 (t)>

The Hilbert transform serves to associate therewith a signal of theform:

    q(t)=A sin(wt+Φ)+n'(t), with <n'.sup.2 (t)>=σ.sup.2

From which it can be deduced that the corresponding analytic signal isof the form:

    x(t)=A exp[j(wt+Φ)]+n*(t), where <|n*(t)|.sup.2 >=2σ.sup.2

The signal to noise ratio ρ of all three signals r(t), q(t), and x(t) isthe same and is given by:

    ρ=A.sup.2 /2σ.sup.2

The analytic signal is determined with periodicity T which is fixed as afunction of the uncertainty Δw with which the angular frequency isknown:

    w=w.sub.O +Δw

By sampling the real signal, it is possible to obtain at each instantt_(n) a phase value Φ_(n) modulo ±π(i.e. lying in the interval -π to+π), i.e.

    x(t)=A exp(jΦ.sub.n)

    t=t.sub.n

    Thus w.sub.O t.sub.n =w.sub.O (t.sub.n -T)+w.sub.O

    Φ.sub.n =(Φ.sub.n-1 +w.sub.O t.sub.n) modulo ±π

Using the following notation:

Δw=the estimated value of the error in the estimated frequency

Φ=the estimated value of the phase

δw=the error in the frequency estimate

δΦ=the error in the phase estimate

It is possible to write Φ_(n) in the form:

    Φ.sub.n =[(Δw)t.sub.n +Φ+(δw)t.sub.n +δΦ]mod ±π

One such signal sample at instant t_(n) and obtained using estimates, isshown in FIG. 2.

in order to estimate the quantities Δw and Φ, it is necessary to developthe phase so as to eliminate the modulo folding and thus reduce theproblem to one of estimating the two characteristic parameters of astraight line.

The sampling period T makes it possible to make a connection withoutambiguity between the phases of two successive samples. If theuncertainty in the angular frequency is Δw, then the sinewave beinganalyzed must rotate through less than π/2 between two successivesamples, and the noise band must be less than 1/2T.

    I<T/(2|Δw|)

This condition is necessary and sufficient. The noise cannot give riseto a phase error of more than ±π/2 which, when added to the range ofuncertainty on the sinewave (±π/2), gives a range of uncertainty whichis less than one turn (±1/2turn).

Thus, using a unit of time equal to the sampling period (so that t is arelative integer), the algorithm for calculating the developed phase maybe described as follows:

Φ_(t) =the developed phase

φ_(t) =the signal phase modulo ±]

ΔΦ=the error in the developed phase between two successive samples.

    ΔΦ=φ.sub.t+1 -φ.sub.t if -π≦φ.sub.t+1 -φ.sub.t <π

    ΔΦ=φ.sub.t+1 -φ.sub.t -2π if φ.sub.t+1 -φ.sub.t <π

    ΔΦ=φ.sub.t+1 -φ2 +2π if φ.sub.t+1 -φ.sub.t <-π

    Φ.sub.t+1 =Φ.sub.t +ΔΦ

FIG. 3 shows how the operation of constructing the developed phase isperformed.

Straight line 10 is a theoretical phase line and:

*=the measured value of the sample phase; and

□=the calculated value of the developed phase (where different form themeasured value).

The expressions described above can be summarized by:

    ΔΦ=(Φ.sub.t+1 -Φ.sub.t) mod ±π

Using the above construction, the developed phases may be explained by:

    Φ.sub.t.sbsb.n =(Δw)t.sub.n +Φ+(δw)t.sub.n +δΦ.sub.t.sbsb.n

which is applicable regardless of t (modulo term omitted).

Non-biased estimators are now used to estimate the frequency and thephase, i.e. estimators such that: <δw>=<δΦ>=0, where δw and δΦ representestimate errors. Assuming that amplitude samples of the real signalsr(t) and q(t) as described above are available, then the sampled phaseat each sampling instant t_(n) is given by:

    φ.sub.t.sbsb.n =Arctan[q(t.sub.n)/r(t.sub.n)]

Work is then performed on the phase samples obtained in this way.Starting from the equation:

    Φ.sub.t.sbsb.n =(Δw)t.sub.n +Φ+(δw)t.sub.n +δΦ.sub.t.sbsb.n

and taking the time origin as being in the middle of the estimationrange T which includes the sample under consideration, the two sides ofthe equation are summed over T and the mathematical expectation of theresult is taken. This gives:

    Φ=(1/l)Σ.sub.t Φ.sub.t.sbsb.n where Σ.sub.t t.sub.n =0

and l is the number of samples taken into account.

multiplying both sides of the same equation by t_(n) prior to summingover T and taking the mathematical expectation then gives Δw:

    Δw=(Σ.sub.t t.sub.n Φ.sub.t.sbsb.n)/(Σ.sub.t t.sub.n.sup.2)

The variance and covariance of each estimated quantity can then becalculated:

    <(Φ-Φ).sup.2)>=(1/l)σ.sub.o.sup.2 =(1/l)(σ.sup.2 /A.sup.2)=1/(2lρ)

where ρA² /σ²

Since the δΦ_(t) are decorrelated therewith and of variance σ_(o) ² =σ²/A²

    <(Δw-Δw).sup.2 >=σ.sub.o.sup.2 /Σ.sub.t t.sub.n.sup.2 ≈12σ.sub.O.sup.2 /l.sup.3 t.sup.2 =(12/l)[σ.sub.o.sup.2 /(l)[σ.sub.o.sup.2 /lT).sup.2 ]

    <(Δw-Δw)(Φ-Φ)>=0

It should be observed that the parameters are directly related either tothe form factor of the observed signal, or else to the signal to noiseratio thereof (ρ).

It can be shown that if the input noise gives rise to phase noise havingGaussian distribution, then these estimators are optimal in the sense oflikelihood maximum and of root mean square error.

As described above, combined use of the analytic signal and ofestimators based on the construction of the developed phase associatedwith the phase of the analytic signal modulo ±πmakes it possible toobtain an estimated value of the phase and of the frequency of theanalyzed signal, given by:

    Φ=(1/l)Σ.sub.t Φ.sub.t.sbsb.n

    Δf=(1/2π)Δw=(1/2π)[Σ.sub.t t.sub.n φ.sub.t.sbsb.n)/(Σ.sub.t t.sub.n.sup.2)]

where Δf is the error relative to the expected frequency f_(O).

Since this estimator is non-biased, an estimate error is obtained whichis zero on average and which has a standard deviation given by:

    (Δw-Δw).sup.2 =(12/l)[σ.sub.o.sup.2 /(lT).sup.2 ]

with

σ_(o) ² =σ_(2/A) ²

σ² =<n² (t)>(noise variance)

l=the number of samples taken into account

T=the sampling period.

These results are valid only if the condition that the noise band of thesignal is less than 1/2T is satisfied.

If this is true, it is possible to constitute a frequency-meter havingthe structure shown in FIG. 5.

Such a frequencymeter comprises a digital processor member 20 having aninput to which the signal to be analyzed SA is applied after passingthrough a digitizer 21 which also receives a reference clock 22 andhaving outputs providing a quality signal and a frequency measurementsignal.

The digital processor member comprises:

a common memory circuit 24;

a circuit 25 for generating the analytic signal;

a circuit 26 for calculating phase samples of the analytic signal andfor constructing the developed phase;

a circuit 27 for monitoring the quality of the signal enablingstatistical estimates to be made of errors using the estimated sinewaveand enabling noise to be estimated as a function of the form of thesignal to be analyzed;

a frequency estimating circuit 28;

with all of the above circuits being interconnected by a both-way linkto an interconnection bus 29;

a digitized sample acquisition circuit connected to the input and alsoconnected to the common memory 24 over a one-way link; and

a coupler 30 whose inputs are connected to the outputs from the qualitymonitor circuit 27 and the frequency-estimating circuit 28, and whoseoutput constitutes the output from said processor member 20.

In the block diagram shown, the signal to be analyzed is sampled using astable reference clock, and is digitized using conventional principles:with analog-to-digital conversion having characteristics (speed,sampling rate, linearity, . . . ) which are critical for obtainingdesired performance in frequency measurement accuracy and in estimatingthe quality of the signal.

The noise specific to digitizing must be negligible compared with theparameters to be acquired.

The samples obtained in this way are acquired and are stored in adigital processor unit having an addressable memory of adequate size forenabling each of the samples required for estimation purposes to beaccessed as many times as desirable by any of the individual processingcircuits.

In accordance with the invention, each of the processing circuits iscompletely independent from the other circuits and can address thememory in order to read therefrom any of the results obtained by theother circuits together with the signal samples originally writtentherein, and also together with its own results, and which can write theresults of its own calculations in a specific zone of said memory whichis attributed to it alone.

In a phasemeter, as shown in FIG. 6, circuits which are identical tothose used in the FIG. 5 frequencymeter have the same referencenumerals.

Said phasemeter includes a second digitizer 31 which receives a phasereference signal SR and the stable reference clock 22.

The digital processor member 40 further includes:

a second digital sample acquisition circuit 33 for acquiring samples ofthe phase reference signal, and for storing them in the same manner asthe first acquisition circuit 23 in the common memory circuit 24; and

a circuit 38 for estimating phase in conjunction with the phasereference signal, and for estimating jitter.

In both of these two devices (frequencymeter, phasemeter), the followingoperations take place successively:

(A) Digitizing the real signal to be analyzed

The signal is digitized using conventional methods performed bycomponents which are available on the market. The characteristicsselected for digitizing should take account of the signal that is to beanalyzed in order to avoid adding error to the resulting estimates.

The main characteristics are as follows:

(a) sampling frequency and anti-aliasing filter.

The selected sampling frequency should have a value F_(e) which iscompatible with the frequency band ΔF_(n) of the noise signalsassociated with the signal to be analyzed, such that:

    F.sub.e >2ΔF.sub.n

The anti-aliasing filter preceding the digitizer should be defined forthe purpose of limiting the band of the signals to be digitized to onehalf of the sampling frequency, with the characteristics within the banddepending on the transfer function of the sampler.

(b) sampler linearity; and

(c) quantification characteristics.

The number of quantification steps and the accuracy and stability of thecorresponding thresholds should have a negligible effect on the finalresults.

(B) Transformation of the samples

The samples of the real signal coming from the digitizer are transformedinto a sampled analytic signal, in particular by calculating samples ofthe quadrature signal from a group of samples of the real signal.

One of the possible methods which may be adopted when using a samplingfrequency which is high relative to the band of the signal to bedigitized is as follows:

Define a sampling frequency which is an integer multiple of a frequencywhich if four times greater than the band of the frequency to beanalyzed (i.e. N times greater).

Take the consecutive samples into account by taking one pair every Nsamples. The first of each pair of samples taken into account isrepresentative of the real signal and the second is representative ofthe quadrature signal (offset by π/2 relative to the initial samplingperiod).

This method requires a sampling frequency which is high relative to theband to be analyzed.

When this condition is not satisfied, it is necessary to interpolate theinput real signals in order to obtain two sampled in quadrature with theequivalent real signal.

The analytic signal can also be stored in the form of complex samples[n*(t)+jq*(t)] appearing at a rate F_(SA) >ΔF_(n) /2, where ΔF_(n)represents the frequency band of the signal being processed.

On the basis of the complex samples of the analytic signal, samples aregenerated of the associated phase and of the developed phase, asdescribed above.

Using the above-described estimators, the following are deducedtherefrom:

the estimated frequency and phase values of a sinewave; and

the variance and the covariance of the estimated quantities using theinput signal as a reference.

Thus, taking the number of samples taken into account as the parameter,the following are obtained;

the value of the frequency;

the value of the phase; and

the estimation noise of the measurement.

This last item can be used to monitor the quality of the measurementcompared with any desired degree of accuracy. This estimation noise sumsup all of the noise regardless of the origin thereof:

noise added to the signal;

non-linearity in the chain to be measured, with non-linearity in themeasuring apparatus being eliminated or at least reduced to negligiblevalues compared with the desired accuracy;

phase noise from the local oscillator in the chain to be measured;

the form factor of the signals to be analyzed (it may be necessary tomeasure rectangular signals, triangular signals, Gaussian signals, etc.)with the resulting noise then being due to harmonics;

estimate errors in the measurement chain related to the estimators.

Since the signal is sinusoidal, the following procedure is used:starting from frequency and phase estimation the theoretical sinusoidalsignal deduced from the estimation is modelled.

Phase variance is proportional to noise and inversely proportional tothe number of samples. The quality of the phase measurement cantherefore be determined from the number of samples taken into accountand the variance of the calculated phase, and the signal-to-noise ratioof the signal being tested can be deduced therefrom:

    A.sup.2 /σ.sup.2 =1/l<(Φ-Φ).sup.2 >

By evaluating the variance in the frequency estimate, it is possible todefine the number of samples that need to be taken into account, forgiven noise σ_(o) ² (see above), in order to be sure of avoidingambiguity in estimate accuracy.

If the shape of the signal is known, the corresponding signal ismodelled with estimated amplitude phase and frequency, and it is thenpossible to proceed as though the signal were a sinewave.

The method of the invention thus makes it possible:

when the noise degrading the signal to be measured is not known, toadjust the duration of the signal taken into account in order to causephase variance to be small and to evaluate:

A² /o² of the signal to be measured;

the corresponding accuracy in the frequency measurement;

the phase value;

the frequency value; and

if the shape of the signal is not known, to model the shape in order toobtain the harmonic content of the signal to be measured and thus deduceother items therefrom;

otherwise, if the type of noise added to the signal and the envelopethereof are known, it is possible to model the noise in order to take itaccount when evaluating other disturbances;

harmonic disturbances;

phase noise; . . . .

Naturally, the present invention has been described and shown purely byway of preferred example, and its component parts could be replaced byequivalent items without thereby going beyond the scope of theinvention.

Any hardware architecture capable of satisfying the following conditionsmay be used for implementing the invention:

tasks are shaped between different calculating members which operateindependently from one another;

all of the above-defined calculating members have read access to all ofthe results of tasks performed by the other members. This type of accessmay be organized, for example, about addressable memories which areaccessible over a common or "multiport" bus, or else using memorieswhich are connected in a lattice network which interconnects the variouscalculating members; and

each of the calculating members has exclusive authority to write in aspecific zone of the memory (or memories).

In this way, the various circuits of devices implementing the method ofthe invention (frequencymeters, phasemeters) may also be replaced bydifferent estimation functions which are implemented in calculatingmembers constituted by software.

We claim:
 1. A method of digitally evaluating at least one desiredparameter of an input signal, said input signal being in the form ofdigitized samples having sample values, wherein the method comprises thesteps of:deriving an analytic signal from said sample values, saidanalytic signal having a real portion which coincides with saiddigitized samples; calculating, in accordance with said sample values,phase values corresponding to the phase of said input signal; estimatingsaid desired parameter by operating on said phase values without usingany operator of the Fourier transform type or any hypothesis test eitherseparately or simultaneously; and estimating the differences between thereal signal as taken in this way and the signal obtained from theestimated parameter, thereby making it possible to deliver data indigital form relating to the quality of the analyzed signal and to thereliability of the estimated values.
 2. A method according to claim 1,wherein said calculating step comprises calculating a developed phase ofsaid analytic signal, and wherein said step of estimating said desiredparameter is performed by operating on said developed phase.
 3. A methodaccording to claim 1, wherein said digitized samples are stored in amemory from which they are extracted as often as may be required by eachnew calculation, and wherein said data in digital form represents, towithin given accuracy, the signal/noise ratio of the input signal, theduration of the input signal to be taken into account, and the accuracyof the evaluation of said desired parameter, with the above being basedon a given quality parameter.
 4. A method according to claim 1, whereinsaid step of estimating said desired parameter is performed in parallelwith said step of deriving said analytic signal.